Age | Commit message (Collapse) | Author | Files | Lines |
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Original commit message from CVS:
all these plugins are moved to gst-plugins-ugly
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Original commit message from CVS:
Port LPCM decoder to 0.9
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Original commit message from CVS:
* configure.ac:
Remove plugins that should have disappeared.
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Original commit message from CVS:
* autogen.sh:
* configure.ac:
Make autogen work again.
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Original commit message from CVS:
* check/elements/audioconvert.c: (setup_audioconvert):
* check/elements/audioresample.c: (setup_audioresample):
* check/elements/volume.c: (setup_volume):
Fix checks.
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Original commit message from CVS:
remove stuff
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Original commit message from CVS:
all these plugins are moved to gst-plugins-good
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Original commit message from CVS:
Ported to GStreamer 0.9. Need to fix performance issues.
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): Use
the second arg for the class, because G_OBJECT_GET_CLASS (self)
returns the wrong thing.
(gst_signal_processor_add_pad_from_template): Make pads of the
right type.
* ext/ladspa/gstladspa.c (gst_ladspa_class_get_param_spec): Make
writable param specs G_PARAM_CONSTRUCT so default values work.
(gst_ladspa_init): Use the second arg for the class.
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE_FULL.
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Original commit message from CVS:
use base class' newsegment to properly timestamp
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compile and register. I have more features tha...
Original commit message from CVS:
2005-08-26 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstladspa.c:
* ext/ladspa/gstladspa.h: Finish porting, still doesn't work but
it does compile and register. I have more features than you.
* ext/ladspa/gstsignalprocessor.h:
* ext/ladspa/gstsignalprocessor.c: Updates, bug fixen.
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Original commit message from CVS:
do proper cleanup/creation, fixes state changes
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Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
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Original commit message from CVS:
some more testing for perfect streams
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Original commit message from CVS:
add a check for audioresample
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Original commit message from CVS:
show some info on what's left in the queue
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transform size protect against possible mem corr...
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
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Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
Revert unpopular change for GST_MESSAGE_SRC to GObject.
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Original commit message from CVS:
add missing files
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Original commit message from CVS:
2005-08-25 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstladspa.h:
* ext/ladspa/gstladspa.c: Halfway-ported. Doesn't compile yet.
* ext/ladspa/gstsignalprocessor.h:
* ext/ladspa/gstsignalprocessor.c: New files, the start of a base
class for DSP elements.
* configure.ac: Sort the external libs checks, add a ladspa check,
output the ladspa makefile.
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Original commit message from CVS:
Fixed EOS and improved robustness for malformed indices.
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Original commit message from CVS:
add lame
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Original commit message from CVS:
fix broken header setup in Makefile.am
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Original commit message from CVS:
dist more
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timestamps.
Original commit message from CVS:
2005-08-24 Andy Wingo <wingo@pobox.com>
* ext/dv/gstdvdemux.c (gst_dvdemux_demux_frame): Send out valid
segment end timestamps.
(Also commit an old changelog entry)
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Original commit message from CVS:
port audioresample to basetransform
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Original commit message from CVS:
enable more; update for basetransform
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Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
GST_MESSAGE_SRC became a GObject
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Original commit message from CVS:
fix speex entry
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Original commit message from CVS:
add speex to spec file and remove gstosslement from POTFILES.in
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Original commit message from CVS:
* ext/speex/gstspeexenc.h:
Fixed include path of adapter
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Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
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Original commit message from CVS:
* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
Fix property warning.
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Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain):
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Small updates, RFC reference to payload encoders.
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otherwise.
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/speex/Makefile.am:
* ext/speex/gstspeex.c: (plugin_init):
* ext/speex/gstspeexdec.c: (speex_get_query_types),
(gst_speex_dec_init), (speex_dec_src_query), (speex_dec_src_event),
(speex_dec_event), (speex_dec_chain):
Port speexdec. Leads to some unfamiliar warnings on console,
but works otherwise.
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opening the mixer.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name
property after opening the mixer.
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Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.c:
* sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
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Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossmixerelement.h:
* sys/oss/gstossmixerelement.c: Added mixer element like
alsamixer.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Register the ossmixer element.
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Original commit message from CVS:
changelog
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Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.h:
* sys/oss/gstosssrc.c: Totally ported, dude.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Add osssrc.
* sys/oss/gstosssink.c: We do native byte order.
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Original commit message from CVS:
Fixed mishandling events and incorrect audio skipping after seek.
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use get_range instead of this seeking nasti...
Original commit message from CVS:
* ext/mad/gstid3tag.c: (gst_id3_tag_init),
(gst_id3_tag_sink_event), (gst_id3_tag_do_caps_nego),
(gst_id3_tag_chain), (gst_id3_tag_change_state), (plugin_init):
Works a bit better now, but still needs a rewrite to use
get_range instead of this seeking nastiness.
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Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/flac/Makefile.am:
* ext/flac/gstflac.c: (plugin_init):
* ext/flac/gstflacdec.c: (flacdec_get_type), (gst_flacdec_init),
(gst_flacdec_update_metadata), (gst_flacdec_seek),
(gst_flacdec_tell), (gst_flacdec_length), (gst_flacdec_read),
(gst_flacdec_write), (gst_flacdec_loop),
(gst_flacdec_get_src_query_types), (gst_flacdec_src_query),
(gst_flacdec_src_event), (gst_flacdec_sink_activate),
(gst_flacdec_sink_activate_pull), (gst_flacdec_change_state):
* ext/flac/gstflacdec.h:
Port flacdec (seeking is still slow'ish).
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Original commit message from CVS:
Fixed some seeking issues
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Original commit message from CVS:
add mpegaudioparse to spec file
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Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
Remove get_time code that is both wrong and unneeded.
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the caps or a base class...
Original commit message from CVS:
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain),
(gst_rtph263penc_set_property), (gst_rtph263penc_get_property):
* gst/rtp/gstrtph263penc.h:
Added configurable pt and ssrc, to be merged in the caps or
a base class...
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Original commit message from CVS:
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_init),
(gst_rtph263pdec_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Some cleanups in the h263p (de)payloaders.
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Original commit message from CVS:
add mpegaudioparse to configure.ac
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